Microsecond resolution is system dependent anyway, so it wasn't reliable.
This CL verifies millisecond timestamps instead of microsecond in tests.
Bug: webrtc:8111
Change-Id: I14aab9a807f747a88b2b84f51becf54f4097931e
Reviewed-on: https://webrtc-review.googlesource.com/c/105561
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25138}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Reland Fix:
- cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional
root level configuration.
- peerconnectionfactory - If this optional is set will now overwrite the
underyling value.
This along with the other field will be deprecated once dependent projects
are updated.
TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org
Bug: webrtc:9681
Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d
Reviewed-on: https://webrtc-review.googlesource.com/c/105560
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25135}
This reverts commit ac2f3d14e45398930bc35ff05ed7a3b9b617d328.
Reason for revert: Breaks downstream project
Original change's description:
> Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h
>
> Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
> that only handles SRTP configuration to a more generic structure that can be
> used and extended for all per peer connection CryptoOptions that can be on a
> given PeerConnection.
>
> Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
> accessed as crypto_options.srtp.whatever_option_name. This is more inline with
> other structures we have in WebRTC such as VideoConfig. As additional features
> are added over time this will allow the structure to remain compartmentalized
> and concerned components can only request a subset of the overall configuration
> structure e.g:
>
> void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
>
> In addition to this it made little sense for sslstreamadapter.h to hold all
> Srtp related configuration options. The header has become loo large and takes on
> too many responsibilities and spilting this up will lead to more maintainable
> code going forward.
>
> This will be used in a future CL to enable configuration options for the newly
> supported Frame Crypto.
>
> Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
> Bug: webrtc:9681
> Reviewed-on: https://webrtc-review.googlesource.com/c/105180
> Reviewed-by: Emad Omara <emadomara@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25130}
TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
Bug: webrtc:9681
Change-Id: Ib0075c477c951b540d4deecb3b0cf8cf86ba0fff
Reviewed-on: https://webrtc-review.googlesource.com/c/105541
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25133}
This patch make CreateOffer/CreateAnswer use the ice credentials
of pooled sessions (if any).
BUG=webrtc:9807
Change-Id: I51e0578f2ff0d4faa93d9666bd6b2c15461e8985
Reviewed-on: https://webrtc-review.googlesource.com/c/102923
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25132}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
Bug: webrtc:9681
Reviewed-on: https://webrtc-review.googlesource.com/c/105180
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25130}
These are not used and removing them makes an upcoming CL moving
the PacketInfo struct much simpler.
Bug: webrtc:9586
Change-Id: I23acb93d9e15f6664e2fa93de744f156546dcbd0
Reviewed-on: https://webrtc-review.googlesource.com/c/105004
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25127}
This adds tracking of traffic for streams that are part of bitrate
allocation but without packet feedback to send side congestion
controller.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: I13e994461c26638d76e8f2f115e6d375e4403116
Reviewed-on: https://webrtc-review.googlesource.com/c/104940
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25126}
And drop a few unneeded includes of rtc_base/stringencode.h.
Bug: webrtc:6424
Change-Id: I8be92a2ca199afaae1d3a177c23acbf2b9bdc465
Reviewed-on: https://webrtc-review.googlesource.com/c/105002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25125}
This prepares for an upcoming CL removing the SocketAdress members.
Bug: webrtc:9586
Change-Id: Iacb03a106f1b143bd2d401a621abb99847a634ed
Reviewed-on: https://webrtc-review.googlesource.com/c/105325
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25124}
In this CL the use of the stationarity properties at init is set to true by default.
Bug: webrtc:9865, chromium:894439
Change-Id: I716ce0d792a50616dc38cc0ba6f2c702549a81cc
Reviewed-on: https://webrtc-review.googlesource.com/c/105303
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25123}
Also remove header extension map from NetEqEventLogInput and RtcEventLogSource.
Bug: webrtc:8111
Change-Id: Ic9be7b03e32ab8aa12284596e21e53b6763f483a
Reviewed-on: https://webrtc-review.googlesource.com/c/102622
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25122}
This CL extend critical sections to incorporate:
* private_submodules_->echo_controller
* config_
As a side benefit, it prevents weird interleaving where configuration
could have been changed in the middle of GetStatistics methods.
Bug: webrtc:9841
Change-Id: I0de5e756a684c2ff1be4effccf8c0f3d3175e3b9
Reviewed-on: https://webrtc-review.googlesource.com/c/105142
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25121}
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."
This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch
It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: I21b4d1fb72710ee3a101888bb6a0b11e0aea35d8
Reviewed-on: https://webrtc-review.googlesource.com/c/105328
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25119}
Since the pushback controller doesn't strictly adhere to the congestion
window, it better belongs together with the congestion controller logic.
Also ensuring that it does not override the configured min bitrate.
Bug: webrtc:9586
Change-Id: I57dcfc946d470247e66c67adabddaafa3d9d83ad
Reviewed-on: https://webrtc-review.googlesource.com/c/105102
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25115}
This CL replaces CHECKs and crashes with DCHECKs and default values.
Bug: webrtc:9535
Change-Id: Ib4b16421699c633d0e9ef140189861c8179450f4
Reviewed-on: https://webrtc-review.googlesource.com/c/105003
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25111}
This CL removes outdated code for testing of platforms with clock-drift
Bug: webrtc:8671
Change-Id: Ie202c514609d9f3d2357107b0daf895331275797
Reviewed-on: https://webrtc-review.googlesource.com/c/105183
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25105}
This CL ensures that the default setting for the usage of stationary signal
properties is not overridden by mistake.
Bug: chromium:894243
Change-Id: I85ab65383ee82b5f3153864da7a0cede7776c146
Reviewed-on: https://webrtc-review.googlesource.com/c/105181
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25104}
The SDP attribute extmap-allow-mixed shows that the client supports
mixing of one- and two-byte header extensions within the same stream.
This is supported on the receive side since CL "Parse two-byte header
extensions", commit 07ba2b9445525da3eabf7c188d631abf32279d98.
Bug: webrtc:7990
Change-Id: I8419da48673f513fcca21a8722614f4601a075fc
Reviewed-on: https://webrtc-review.googlesource.com/c/103420
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25098}
Basic integration of media_transport in JSepTransportController.
- Creates media_transport if media transport factory provided in jsep transport controller configuration.
- Unittest that makes sure media_transport is created with correct caller or callee setting.
- Added fake_media_transport, for now simple implementation which only stores caller/callee, but in the future fake media transport will be expanded to pass frames between two fake media_transports, which will enable audio / video integration tests.
NEXT STEPS: Once integration of media_transport with PeerConnection (https://webrtc-review.googlesource.com/c/src/+/103860) lands, we can start passing media transport factory from peer connection to jsep transport controller.
NOTE: Includes missing include change from https://webrtc-review.googlesource.com/c/src/+/103540 (otherwise this change will not compile)
Bug: webrtc:9719
Change-Id: I1e8a521beab445aa9f7ea93c8d7a537dc137d11c
Reviewed-on: https://webrtc-review.googlesource.com/c/104400
Commit-Queue: Anton Sukhanov <sukhanov@google.com>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25096}
This change prevents resets unless someone actually set a FrameEncryptor
/ FrameDecryptor.
Bug: webrtc:9795
Change-Id: I29910b9ecc2f6f8eea371c5961ac7e9780de65d2
Reviewed-on: https://webrtc-review.googlesource.com/c/104901
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25095}
After landing https://webrtc-review.googlesource.com/c/src/+/104802, it
is finally possible to remove the complexity behind
rtc_base:rtc_base_approved and switch back to one build target.
The long term vision is to remove it too, in favor of smaller and more
focues build targets.
Bug: webrtc:9838
Change-Id: Ib98dfae103a20edb8c8b6706d376ad4f3c992886
Reviewed-on: https://webrtc-review.googlesource.com/c/105041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25093}
compare_videos.py will now print the VMAF score for each frame.
The CL also removes some stale comments.
Bug: webrtc:9642
Change-Id: I5623588580dea06dd487d7763dc3a2511bd2cd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/105103
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25092}
The logic in rtc::Buffer::OnMovedFrom was backwards w.r.t.
RTC_DCHECK_IS_ON. We intended to provoke bugs when DCHECKs are on and
play it safe when DCHECKs are off, but actually we did the reverse.
This CL fixes that.
It also adds a death test that would have caught the bug.
Bug: webrtc:9856
Change-Id: Ib6a4b07d12732e5a66e93b36b885abdab93e55c7
Reviewed-on: https://webrtc-review.googlesource.com/c/105040
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25091}
I don't think this has any impact, just wanted to have a first unit
test to play around with.
Bug: None
Change-Id: I892e2642f0243c5c9ee807cf71abcd96112b25f4
Reviewed-on: https://webrtc-review.googlesource.com/c/105000
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25089}
